Hi all!
I’m brand new to FreeSwitch, but have some
experience with SipX.
Our company is evaluating FS.
For test purposes I set up FS on a virtual machine (vmware
esxi). I use CentOS.
The FS version I use is 1.0.5-20100110-0400.
I have a question regarding sip trunk between FS and
SipX.
I created the following GW in external profile:
[freeswitch@freeswitch external]$ cat sipx-lab.xml |
grep -v '<!--'
<include>
<gateway
name="sipx4.lab.nstel.ru">
<param name="username"
value="zxcv"/>
<param name="password"
value="2007"/>
<param name="register"
value="false"/>
</gateway>
</include>
External.xml file is not modified after installation.
I see this gateway via fs_cli:
freeswitch@internal>
=================================================================================================
Name
sipx4.lab.nstel.ru
Profile external
Scheme
Digest
Realm
sipx4.lab.nstel.ru
Username
zxcv
Password
yes
From
<sip:zxcv@sipx4.lab.nstel.ru;transport=udp>
Contact
<sip:gw+sipx4.lab.nstel.ru@172.23.22.49:5080;transport=udp;gw=sipx4.lab.nstel.ru>
Exten
zxcv
To
sip:zxcv@sipx4.lab.nstel.ru
Proxy
sip:sipx4.lab.nstel.ru
Context
public
Expires
3600
Freq
3600
Ping
0
PingFreq 0
PingState 0/0/0
State
NOREG
Status
UP
CallsIN
0
CallsOUT 0
=================================================================================================
I created new FS extension 2853. I registers (xlite)
and I can call it.
Now I want to call FS user from sipx.
From the sipx side one can configure link to FS
differently. There are two options:
1. Call FS directly trough sipxproxy (the core part
of sipx works as sip proxy, not as b2bua)
2. Call trough embedded b2bua, named sipxbridge.
When a call is going trough sipxbridge, it is
successfully landed at FS extension 2853.
But when a call is going from the sipxproxy, it is
challenged with Status 407 “proxy authentication required’, and
then call fails.
(I’m not sure if sipx should handle this
challenge, but this is separate question for the sipx forum).
At the default log level I see nothing in the
freeswitch.log.
So the question is why one call through external
profile is being challenged while the other is not?
I suspect that the reason is in the difference in the
two Invite messages:
Here is challenged Invite:
INVITE sip:2853@fs.lab.nstel.ru:5080;transport=udp
SIP/2.0
Record-Route:
<sip:172.23.12.104:5060;lr;sipXecs-rs=%2Aauth%7E.%2Afrom%7EOTRiZjI3MjgtYWMxNzBjZTktMTNjNC0yZjZkZWMtNjNkNmUwN2UtMmY2ZGVj.900_ntap%2Aid%7EMjIyNjAtMQ%60%60%213ba6d85f6946c4c6001bee1d3b54474f>
From:
"testphone3857"<sip:3857@lab.nstel.ru>;tag=94bf2728-ac170ce9-13c4-2f6dec-63d6e07e-2f6dec
To: <sip:2853@lab.nstel.ru>
Call-Id:
94bed2a0-ac170ce9-13c4-2f6dec-68ffdb81-2f6dec@lab.nstel.ru
Cseq: 2 INVITE
Via: SIP/2.0/UDP 172.23.12.104;branch=z9hG4bK-sipXecs-000eaf48a36fe4029c7cde004a6f44424847
Via: SIP/2.0/TCP
172.23.12.104;branch=z9hG4bK-sipXecs-000bf3008cfa3f3e70470789c75232ba9499~ac7fd729330fd563f83cacd941311e75;id=22260-1
Via: SIP/2.0/UDP
172.23.12.233:5060;branch=z9hG4bK-2f6dec-b9456412-7ad07784
Max-Forwards: 18
Supported: replaces
User-Agent: LG-Nortel LIP 6812 v1.2.38sp
SN/00405A18B634
Contact:
<sip:3857@172.23.12.233:5060;x-sipX-nonat>
Proxy-Authorization: Digest
username="3857",realm="lab.nstel.ru",nonce="e2152722611af1bbe59f3a8eb31b8eb84b4db2ca",uri="sip:2853@lab.nstel.ru",response="a09c34fbd897783c9b9af50bd044ccaa",algorithm=MD5
Content-Type: application/sdp
Content-Length: 301
Date: Wed, 13 Jan 2010 11:47:22 GMT
Expires: 60
X-Sipx-Handled: X172.23.12.104-81.211.30.104
v=0
o=LGEIPP 16246 16247 IN IP4 172.23.12.233
s=SIP Call
c=IN IP4 172.23.12.233
t=0 0
m=audio 23008 RTP/AVP 0 8 18 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexb=no
a=fmtp:101 0-11
a=sendrecv
And here is successful Invite:
INVITE sip:2853@fs.lab.nstel.ru;user=phone SIP/2.0
Call-ID:
94bec3b8-ac170ce9-13c4-2f6d47-2fee09d0-2f6d47@lab.nstel.ru.0
CSeq: 1 INVITE
From: "testphone3857"
<sip:3857@172.23.12.104>;tag=816640414244159961
To: <sip:2853@fs.lab.nstel.ru;user=phone>
Via: SIP/2.0/UDP
172.23.12.104:5080;branch=z9hG4bK885dde2c8680f5315845cd3350b8b605373534
Max-Forwards: 70
User-Agent: sipXecs/4.0.2 sipXecs/sipxbridge (Linux)
P-Asserted-Identity: <sip:1004@172.23.12.104>
Contact:
<sip:1004@172.23.12.104:5080;transport=udp>
Route: <sip:172.23.22.49:5080;transport=udp;lr>
Session-Expires: 1800;refresher=uac
Allow: INVITE,BYE,ACK,CANCEL,OPTIONS
Content-Type: application/sdp
Content-Length: 315
v=0
o=sipxbridge 6640787141824452741 1 IN IP4
172.23.12.104
s=SIP Call
c=IN IP4 172.23.12.104
t=0 0
m=audio 30248 RTP/AVP 0 8 18 4 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:18 annexb=no
a=fmtp:101 0-11
a=sendrecv
Can somebody please if it is a FS configuration
problem or a software problem or is it a problem on the sipx side?
Thanks in advance,
Nikolay.